On Tue, Aug 5, 2008 at 12:37 PM, Tzury Bar Yochay <tzury.by at gmail.com>wrote: > Hi, > > I am trying to make a (decent) VoIP call using the following setup: > - Asterix PBX (latest version) running on Trixbox > - pjsua (command line tool) on a Windows XP host PC > - Cellphone (O2) running Windows Mobile 5 and pjsua_wince application > FYI PocketPJ is the new (and better) app sample for WM. > > - LAN is connected using a LinkSys WRT54L (PC connected with cable and > phone connected with WiFi) > > First thing first, which pjsip version are you using on both sides of the call? > I manage to make a phone between the O2 and the PC, however the quality is > very poor, with echo and disruptions of the voice. > This probably is a known problem. We use echo suppressor on WM and it's performance is quite poor. If anyone can suggest better alternatives then that will be great. > Also, I periodically (~1 sec interval) get "silence noises" when there is > no talk. > > I suppose that's the VAD and keep-alive in action. This is also a known issue: http://trac.pjsip.org/repos/ticket/490 > dq dump from pjsua: > [CONFIRMED] To: "801" <sip:801 at 10.0.0.151 <sip%3A801 at 10.0.0.151> > >;tag=as18eb6d76 > Call time: 00h:01m:12s, 1st res in 0 ms, conn in 0ms > SRTP status: Not active Crypto-suite: (null) > #0 PCMU @8KHz, sendrecv, peer=10.0.0.151:10406 > RX pt=0, stat last update: 00h:00m:02.781s ago > total 3.5Kpkt 570.8KB (713.6KB +IP hdr) @avg=63.1Kbps/78.8Kbps > pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) > (msec) min avg max last dev > loss period: 0.000 0.000 0.000 0.000 0.000 > jitter : 0.000 3.517 21.375 4.000 2.393 > TX pt=0, ptime=20ms, stat last update: 00h:00m:02.375s ago > total 3.5Kpkt 575.4KB (719.3KB +IP hdr) @avg 63.6Kbps/79.5Kbps > pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) > (msec) min avg max last dev > loss period: 0.000 0.000 0.000 0.000 0.000 > jitter : 18.875 21.696 24.375 22.375 1.661 > RTT msec : 2.014 5.922 11.016 5.020 2.840 > > > I have the following questions: > 1. Why is the PCMU codec selected and not Speex (I have set Speex to be in > the highest priority on both sides)? How can I make it choose something else > other than PCMU? > When I tries setting all Codecs priorities except Speex to zero, I couldn't > establish a conversation. > > Did you put/enable Speex in the build? > 2. I set --no-vad in the pjsua console on the PC and set: > # define PJMEDIA_CODEC_MAX_SILENCE_PERIOD -1 > before deploying the file to the mobile device. > This should have disabled the 'silence noises' but it doesn't. Am I missing > something? > > Disabling the vad with --no-vad should be sufficient. Of course you'd need to do the same in the WM side as well. > 3. Although I use non-routable IPs, I have the entire system on a LAN. Why > does it think I'm behind a NAT and tunnels everything through the Asterix? > Is there a way to tell the software to contact directly? > I've set the following in pjsua_wince.cpp: > // Use this to disable STUN > # define STUN_DOMAIN NULL > # define STUN_SERVER NULL > And it does not seem to have any effect. > > If you're calling an Asterisk number (or otherwise calling via Asterisk), then it's up to your Asterisk configuration whether to tunnel the audio or not, no? > 4. Was anyone able to succesfully deploy such (or similar) enviorment? Can > you provide some hints? > > 5. It seems that the changes I make have very little or no effect on the > overall quality. For example, I set: > # define PJMEDIA_SOUND_BUFFER_COUNT 20 > and expected to have a very long delay, but it had no effect. > Am I doing something wrong? > > With the latest (0.9) pjsip, that setting has less effect now as the buffer count will be calculated dynamically. Cheers Benny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080805/b551661b/attachment-0001.html
